[asterisk-dev] [Code Review] 3474: TLS and SRTP status not available in CLI

[Olle E Johansson]

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A really good addition. I think this has to go in to all releases. Working on related issues in my juniper branch on spare time, to expose more information about the actual TLS session, like TLS/SSL version, crypto used etc. 


/branches/11/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/3474/#comment21499>

    If the header is just "signalling" we should also list other transports than TLS - UDP, TCP, WS, WSS. "non-TLS" is not a good solution :-)


- Olle E Johansson


On April 23, 2014, 9:52 p.m., Patrick Laimbock wrote:
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3474/
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> (Updated April 23, 2014, 9:52 p.m.)
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> 
> Review request for Asterisk Developers.
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> Bugs: ASTERISK-23564
>     https://issues.asterisk.org/jira/browse/ASTERISK-23564
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> Repository: Asterisk
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> Description
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> AFAICT there is no way to see from the CLI if TLS and SRTP are enabled for a channel. I asked on the ML and in #asterisk but received no answer other than that nobody knew how to get that info from the CLI. This patch shows TLS or non-TLS and SRTP or RTP.
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> Diffs
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>   /branches/11/channels/chan_sip.c 412921 
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> Diff: https://reviewboard.asterisk.org/r/3474/diff/
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> Testing
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> Testing was done on Asterisk-11.8.1 with TLS & RPT, TLS & SRTP, non-TLS & RPT configured and a Nexus GSM using Linphone with similar configs. AFAICT the status of the channel and media was correctly reported for each scenario.
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> Thanks,
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> Patrick Laimbock
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>

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